1. Technical Field
The present disclosure relates to network communications, and more particularly to a volume control method and system typically applied to a Voice over Internet Protocol (VOIP) system.
2. Description of Related Art
FIG. 1 is a block diagram of a structure of a typical volume control system applied to an audio processing system, for example, a VOIP system. The volume control system comprises a transmitter (an outgoing side) 11 and a receiver (an incoming side) 13, each comprising a plurality of software and/or hardware modules or components.
The transmitter 11 at least comprises an analog-to-digital converter (ADC) 112, a codecing unit 114 and a packetizing module 116. The receiver 13 at least comprises a gain control unit 130, a digital-to-analog converter (DAC) 132, a codecing unit 134 and a de-packetizing module 136.
The ADC 112 retrieves an analog audio stream from an audio resource (not shown), and converts the analog audio stream to a digital audio stream. The codecing unit 114 samples the digital audio stream to generate a plurality of digital audio frames, and encodes (compresses) the sampled digital audio frames according to G.711 codecing rules. When the encoding process is complete, the codecing unit 114 outputs the encoded digital audio frames to the packetizing module 116.
The packetizing module 116 adds an Internet Protocol (IP) header to one of the encoded digital audio frames, and packetizes the encoded digital audio frame into a User Datagram Protocol (UDP) packet using a UDP module (not shown) of the volume control system. In addition, the packetizing module 116 packetizes the UDP packet into a Real-time Transport Protocol (RTP) packet using an RTP module (not shown) of the volume control system, and packetizes the RTP packet into an IP packet.
The above operations are repeated to packetize all of the encoded digital audio frames into IP packets. The packetizing module 116 may also packetize a number of the encoded digital audio frames into a single IP packet.
Each of the IP packets comprises a destination address and, consequently, can be transmitted to a correct destination via an IP network.
The IP packets forming the digital audio stream are transmitted from the transmitter 11 to the receiver 13. The de-packetizing module 136 retrieves and de-packetizes the IP packets into the RTP packets, de-packetizes the RTP packets into the UDP packets using the RTP module (not shown), and de-packetizes the UDP packets into the encoded digital audio frames using the UDP module (not shown).
The codecing unit 134 decodes the encoded digital audio frames to recover the digital audio stream with a pulse code modulation (PCM) format. The DAC 132 converts the digital audio stream to the analog audio stream (audio signals) for broadcast to a user located at the receiver 13. The volume of the audio signals may be controlled by the user with the gain control unit 130.
When a multiparty voice conference is held via a typical audio processing system, volume inconsistency usually occurs at separate incoming sides. Therefore users at each incoming side must manually achieve volume control, by gain control or by adjustment of parameters including linearity for power amplifier noise, for example. It is inconvenient for users to frequently use the volume control and, thus, an improved volume control method and system are desired.